Hi list: i have an asterisk box behind the NAT ,when i try to send calls through Sip to the voip provider server the call is answered but in a one way calling,I hear the voice of the other side just for 4 seconds and then stop but the call do not hangup. my sip.conf is: [voip provider] type=peer host=213.112.50.8 username=XXXXXXX secret=XXXXXX fromuser=XXXXXXX canreinvite=no nat=yes insercure=invite disallow=all allow=gsm __________________________________________ Yahoo! DSL ? Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com
what type of NAT do you have? sync? full cone? cone restricted, port restricted? any messages in asterisk verbose console? best regards On 12/7/05, chawki hammoud <cyhammoud@yahoo.com> wrote:> > Hi list: > i have an asterisk box behind the NAT ,when i try to > send calls through Sip to the voip provider server the > call is answered but in a one way calling,I hear the > voice of the other side just for 4 seconds and then > stop but the call do not hangup. > > my sip.conf is: > [voip provider] > type=peer > host=213.112.50.8 > username=XXXXXXX > secret=XXXXXX > fromuser=XXXXXXX > canreinvite=no > nat=yes > insercure=invite > disallow=all > allow=gsm > > > > > __________________________________________ > Yahoo! DSL ? Something to write home about. > Just $16.99/mo. or less. > dsl.yahoo.com > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051207/6e692a62/attachment.htm
If your Astersik server behind NAT too, your need modify SIP.conf like this.... externalIP= x.x.x.x localnet= x.x.x. hope this can help you.... On 12/8/05, Moises Silva <moises.silva@gmail.com> wrote:> > what type of NAT do you have? sync? full cone? cone restricted, port > restricted? > any messages in asterisk verbose console? > > best regards > > On 12/7/05, chawki hammoud <cyhammoud@yahoo.com> wrote: > > > > Hi list: > > i have an asterisk box behind the NAT ,when i try to > > send calls through Sip to the voip provider server the > > call is answered but in a one way calling,I hear the > > voice of the other side just for 4 seconds and then > > stop but the call do not hangup. > > > > my sip.conf is: > > [voip provider] > > type=peer > > host=213.112.50.8 > > username=XXXXXXX > > secret=XXXXXX > > fromuser=XXXXXXX > > canreinvite=no > > nat=yes > > insercure=invite > > disallow=all > > allow=gsm > > > > > > > > > > __________________________________________ > > Yahoo! DSL ? Something to write home about. > > Just $16.99/mo. or less. > > dsl.yahoo.com > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com<http://easynews.com/>-- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > "Su nombre es GNU/Linux, no solamente Linux, mas info en > http://www.gnu.org" > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>-- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Jeffery -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051207/b8c66c2e/attachment.htm
Hi: i added these two lines to my general context ,but nothing happened the same result the sound came in one way for 3 seconds and stopped but it didnt hangup. --- Jeffery Chen <jeffery9@gmail.com> wrote:> If your Astersik server behind NAT too, your need > modify SIP.conf like > this.... > > externalIP= x.x.x.x > localnet= x.x.x. > > hope this can help you.... > > > On 12/8/05, Moises Silva <moises.silva@gmail.com> > wrote: > > > > what type of NAT do you have? sync? full cone? > cone restricted, port > > restricted? > > any messages in asterisk verbose console? > > > > best regards > > > > On 12/7/05, chawki hammoud <cyhammoud@yahoo.com> > wrote: > > > > > > Hi list: > > > i have an asterisk box behind the NAT ,when i > try to > > > send calls through Sip to the voip provider > server the > > > call is answered but in a one way calling,I hear > the > > > voice of the other side just for 4 seconds and > then > > > stop but the call do not hangup. > > > > > > my sip.conf is: > > > [voip provider] > > > type=peer > > > host=213.112.50.8 > > > username=XXXXXXX > > > secret=XXXXXX > > > fromuser=XXXXXXX > > > canreinvite=no > > > nat=yes > > > insercure=invite > > > disallow=all > > > allow=gsm > > > > > > > > > > > > > > > __________________________________________ > > > Yahoo! DSL ??? Something to write home about. > > > Just $16.99/mo. or less. > > > dsl.yahoo.com > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by > Easynews.com<http://easynews.com/>-- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > > > > > > > -- > > "Su nombre es GNU/Linux, no solamente Linux, mas > info en > > http://www.gnu.org" > > _______________________________________________ > > --Bandwidth and Colocation provided by > Easynews.com <http://easynews.com/>-- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > > > > > -- > Jeffery > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com > -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users>__________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
can u paste your sip.conf general section,,? there have another possible cause... the both side use different codecm and asterisk can not translaste it ... -- Jeffery On 12/8/05, chawki hammoud <cyhammoud@yahoo.com> wrote:> > Hi: > i added these two lines to my general context ,but > nothing happened the same result the sound came in one > way for 3 seconds and stopped but it didnt hangup. > > --- Jeffery Chen <jeffery9@gmail.com> wrote: > > > If your Astersik server behind NAT too, your need > > modify SIP.conf like > > this.... > > > > externalIP= x.x.x.x > > localnet= x.x.x. > > > > hope this can help you.... > > > > > > On 12/8/05, Moises Silva <moises.silva@gmail.com> > > wrote: > > > > > > what type of NAT do you have? sync? full cone? > > cone restricted, port > > > restricted? > > > any messages in asterisk verbose console? > > > > > > best regards > > > > > > On 12/7/05, chawki hammoud <cyhammoud@yahoo.com> > > wrote: > > > > > > > > Hi list: > > > > i have an asterisk box behind the NAT ,when i > > try to > > > > send calls through Sip to the voip provider > > server the > > > > call is answered but in a one way calling,I hear > > the > > > > voice of the other side just for 4 seconds and > > then > > > > stop but the call do not hangup. > > > > > > > > my sip.conf is: > > > > [voip provider] > > > > type=peer > > > > host=213.112.50.8 > > > > username=XXXXXXX > > > > secret=XXXXXX > > > > fromuser=XXXXXXX > > > > canreinvite=no > > > > nat=yes > > > > insercure=invite > > > > disallow=all > > > > allow=gsm > > > > > > > > > > > > > > > > > > > > __________________________________________ > > > > Yahoo! DSL ? Something to write home about. > > > > Just $16.99/mo. or less. > > > > dsl.yahoo.com > > > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by > > Easynews.com<http://easynews.com/>-- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > > -- > > > "Su nombre es GNU/Linux, no solamente Linux, mas > > info en > > > http://www.gnu.org" > > > _______________________________________________ > > > --Bandwidth and Colocation provided by > > Easynews.com <http://easynews.com/>-- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > -- > > Jeffery > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com > > -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051208/e1ce0ff3/attachment.htm
Forward UDP Ports 10000-20000 to your asterisk box. On 12/8/05, Jeffery Chen <jeffery9@gmail.com> wrote:> > can u paste your sip.conf general section,,? > > there have another possible cause... the both side use different codecm > and asterisk can not translaste it ... > > > -- > Jeffery > > > On 12/8/05, chawki hammoud <cyhammoud@yahoo.com> wrote: > > > > Hi: > > i added these two lines to my general context ,but > > nothing happened the same result the sound came in one > > way for 3 seconds and stopped but it didnt hangup. > > > > --- Jeffery Chen <jeffery9@gmail.com> wrote: > > > > > If your Astersik server behind NAT too, your need > > > modify SIP.conf like > > > this.... > > > > > > externalIP= x.x.x.x > > > localnet= x.x.x. > > > > > > hope this can help you.... > > > > > > > > > On 12/8/05, Moises Silva < moises.silva@gmail.com> > > > wrote: > > > > > > > > what type of NAT do you have? sync? full cone? > > > cone restricted, port > > > > restricted? > > > > any messages in asterisk verbose console? > > > > > > > > best regards > > > > > > > > On 12/7/05, chawki hammoud <cyhammoud@yahoo.com> > > > wrote: > > > > > > > > > > Hi list: > > > > > i have an asterisk box behind the NAT ,when i > > > try to > > > > > send calls through Sip to the voip provider > > > server the > > > > > call is answered but in a one way calling,I hear > > > the > > > > > voice of the other side just for 4 seconds and > > > then > > > > > stop but the call do not hangup. > > > > > > > > > > my sip.conf is: > > > > > [voip provider] > > > > > type=peer > > > > > host=213.112.50.8 > > > > > username=XXXXXXX > > > > > secret=XXXXXX > > > > > fromuser=XXXXXXX > > > > > canreinvite=no > > > > > nat=yes > > > > > insercure=invite > > > > > disallow=all > > > > > allow=gsm > > > > > > > > > > > > > > > > > > > > > > > > > __________________________________________ > > > > > Yahoo! DSL ? Something to write home about. > > > > > Just $16.99/mo. or less. > > > > > dsl.yahoo.com > > > > > > > > > > _______________________________________________ > > > > > --Bandwidth and Colocation provided by > > > Easynews.com<http://easynews.com/>-- > > > > > > > > > > Asterisk-Users mailing list > > > > > To UNSUBSCRIBE or update options visit: > > > > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > > > > > > > -- > > > > "Su nombre es GNU/Linux, no solamente Linux, mas > > > info en > > > > http://www.gnu.org" > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by > > > Easynews.com <http://easynews.com/>-- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > > > > > > > -- > > > Jeffery > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com > > > -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051208/55765dbd/attachment.htm
> i have an asterisk box behind the NAT ,when i try to > send calls through Sip to the voip provider server the > call is answered but in a one way calling,I hear the > voice of the other side just for 4 seconds and then > stop but the call do not hangup.SOmetimes this can be due to the client using silence suppression. Make sure this function if OFF.
On 12/8/05, chawki hammoud <cyhammoud@yahoo.com > wrote:> Hi: > > i added these two lines to my general context ,but > nothing happened the same result the sound came in one > way for 3 seconds and stopped but it didnt hangup. > > --- Jeffery Chen <jeffery9@gmail.com> wrote: > > > If your Astersik server behind NAT too, your need > > modify SIP.conf like > > this.... > > > > externalIP= x.x.x.x > > localnet= x.x.x. > > > > hope this can help you....Make sure that you have ports 5060 and ports 10000-20000 UDP forwarded to your Asterisk server. (Asterisk uses UDP for SIP, not TCP!!!) Also, in addition to the externip and localnet entries in sip.conf, You need to add a "nat=yes" entry Tom -------------------- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 "Intelligent technology solutions for small businesses."
It can also be that the NAT is not truly SIP aware as it will create some confusion if the IP address in the IP header is converted, while the IP address in the SIP header is not. One cause would be that messages are send to wrong address. Jan Wilson Pickett wrote:>>i have an asterisk box behind the NAT ,when i try to >>send calls through Sip to the voip provider server the >>call is answered but in a one way calling,I hear the >>voice of the other side just for 4 seconds and then >>stop but the call do not hangup. >> >> > >SOmetimes this can be due to the client using silence suppression. >Make sure this function if OFF. >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051212/d78fe887/attachment.htm